Searched defs:packet (Results 226 - 250 of 509) sorted by path

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/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/
H A Djitter_buffer_unittest.cc19 #include "webrtc/modules/video_coding/main/source/packet.h"
144 VCMPacket packet; local
145 packet.dataPtr = data_buffer_;
146 bool packet_available = stream_generator_->PopPacket(&packet, index);
151 return jitter_buffer_->InsertPacket(packet, &retransmitted);
155 VCMPacket packet; local
156 packet.dataPtr = data_buffer_;
157 bool packet_available = stream_generator_->GetPacket(&packet, index);
162 return jitter_buffer_->InsertPacket(packet, &retransmitted);
268 // Insert the packet t
1863 VCMPacket packet; local
1899 VCMPacket packet; local
1905 VCMPacket packet; local
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H A Dreceiver.cc74 int32_t VCMReceiver::InsertPacket(const VCMPacket& packet, argument
77 // Insert the packet into the jitter buffer. The packet can either be empty or
80 const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
93 timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
267 // dual decoder has caught up with the decoder decoding with packet losses.
H A Dreceiver_unittest.cc15 #include "webrtc/modules/video_coding/main/source/packet.h"
45 VCMPacket packet; local
46 packet.dataPtr = data_buffer_;
47 bool packet_available = stream_generator_->GetPacket(&packet, index);
52 return receiver_.InsertPacket(packet, 640, 480);
56 VCMPacket packet; local
57 packet.dataPtr = data_buffer_;
58 bool packet_available = stream_generator_->PopPacket(&packet, index);
62 return receiver_.InsertPacket(packet, kWidth, kHeight);
74 // Drop the second packet
75 VCMPacket packet; local
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H A Dsession_info.cc13 #include "webrtc/modules/video_coding/main/source/packet.h"
126 VCMPacket& packet = *packet_it; local
129 // Calculate the offset into the frame buffer for this packet.
134 // Set the data pointer to pointing to the start of this packet in the
136 const uint8_t* packet_buffer = packet.dataPtr;
137 packet.dataPtr = frame_buffer + offset;
143 if (packet.codecSpecificHeader.codec == kRtpVideoH264 &&
144 packet.codecSpecificHeader.codecHeader.H264.stap_a) {
147 while (nalu_ptr < packet_buffer + packet.sizeBytes) {
150 length + (packet
438 InsertPacket(const VCMPacket& packet, uint8_t* frame_buffer, VCMDecodeErrorMode decode_error_mode, const FrameData& frame_data) argument
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H A Dsession_info_unittest.cc15 #include "webrtc/modules/video_coding/main/source/packet.h"
176 // Insert empty packet which will be the new high sequence number.
349 // Insert an older packet with a first packet set.
427 // Insert out of bound regular packets, and then the first and last packet.
438 // Insert an older packet with a first packet set.
494 VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(), local
496 EXPECT_EQ(session_.InsertPacket(*packet,
501 delete packet;
551 VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(), local
622 VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(), local
693 VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(), local
765 VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(), local
836 VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(), local
936 VCMPacket* packet = new VCMPacket(packet_buffer_, packet_buffer_size(), local
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/test/
H A Dstream_generator.cc18 #include "webrtc/modules/video_coding/main/source/packet.h"
69 VCMPacket packet; local
70 packet.seqNum = sequence_number;
71 packet.timestamp = timestamp;
72 packet.frameType = type;
73 packet.isFirstPacket = first_packet;
74 packet.markerBit = marker_bit;
75 packet.sizeBytes = size;
76 packet.dataPtr = packet_buffer;
77 if (packet
86 PopPacket(VCMPacket* packet, int index) argument
96 GetPacket(VCMPacket* packet, int index) argument
105 NextPacket(VCMPacket* packet) argument
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/
H A Dmt_test_common.cc55 // Insert outgoing packet into list
61 // Simulate receive time = network delay + packet jitter
77 RtpPacket* packet = NULL; local
83 // Take first packet in list
84 packet = _rtpPackets.front();
85 int64_t timeToReceive = packet->receiveTime - now;
96 if (!parser->Parse(packet->data, packet->length, &header)) {
97 delete packet;
105 if (!rtp_receiver_->IncomingRtpPacket(header, packet
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H A Drtp_player.cc95 void AddPacket(RawRtpPacket* packet) { argument
96 assert(packet);
97 printf("Throw: %08x:%u\n", packet->ssrc(), packet->seq_num());
100 fprintf(debug_file_, "%u Lost packet: %u\n", loss_count_,
101 packet->seq_num());
103 packets_.push_back(packet);
112 RawRtpPacket* packet = *it; local
113 if (ssrc == packet->ssrc() && resendSeqNum == packet
129 RawRtpPacket* packet = *it; local
151 LogPacketResent(RawRtpPacket* packet) argument
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H A Dtest_callbacks.cc64 // will call the VCMReceiver input packet
241 // Take first packet in list
265 // Insert outgoing packet into list
271 // Simulate receive time = network delay + packet jitter
282 RtpPacket* packet = NULL; local
286 // Take first packet in list
287 packet = _rtpPackets.front();
288 int64_t timeToReceive = packet->receiveTime - now;
300 if (!parser->Parse(packet->data, packet
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/external/chromium_org/third_party/webrtc/test/
H A Dfake_network_pipe.cc65 // The packet data.
69 // The time the packet was sent out on the network.
71 // The time the packet should arrive at the reciver.
114 // Too many packet on the link, drop this one.
133 NetworkPacket* packet = new NetworkPacket(data, data_length, time_now, local
135 capacity_link_.push(packet);
163 // Time to get this packet.
164 NetworkPacket* packet = capacity_link_.front(); local
169 delete packet;
174 // earlier than the last packet i
193 NetworkPacket* packet = delay_link_.front(); local
204 NetworkPacket* packet = packets_to_deliver.front(); local
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H A Dnull_transport.cc15 bool NullTransport::SendRtp(const uint8_t* packet, size_t length) { argument
19 bool NullTransport::SendRtcp(const uint8_t* packet, size_t length) { argument
H A Drtcp_packet_parser.cc21 const uint8_t* packet = static_cast<const uint8_t*>(data); local
22 RTCPUtility::RTCPParserV2 parser(packet, len, true);
H A Drtp_file_reader_unittest.cc32 test::RtpFileReader::Packet packet; local
34 while (rtp_packet_source_->NextPacket(&packet))
62 test::RtpFileReader::Packet packet; local
63 while (rtp_packet_source_->NextPacket(&packet))
70 test::RtpFileReader::Packet packet; local
71 while (rtp_packet_source_->NextPacket(&packet)) {
72 RtpUtility::RtpHeaderParser rtp_header_parser(packet.data, packet.length);
/external/chromium_org/third_party/webrtc/video/
H A Dcall.cc104 virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
110 DeliveryStatus DeliverRtcp(const uint8_t* packet, size_t length);
111 DeliveryStatus DeliverRtp(const uint8_t* packet, size_t length);
346 PacketReceiver::DeliveryStatus Call::DeliverRtcp(const uint8_t* packet, argument
349 // Do NOT broadcast! Also make sure it's a valid packet.
351 // there's no receiver of the packet.
359 if (it->second->DeliverRtcp(packet, length))
370 if (it->second->DeliverRtcp(packet, length))
377 PacketReceiver::DeliveryStatus Call::DeliverRtp(const uint8_t* packet, argument
383 const uint8_t* ptr = &packet[
397 DeliverPacket(const uint8_t* packet, size_t length) argument
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H A Dcall_perf_tests.cc64 virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
65 RTCPUtility::RTCPParserV2 parser(packet, length, true);
72 const RTCPUtility::RTCPPacket& packet = parser.Packet(); variable
74 packet.SR.NTPMostSignificant,
75 packet.SR.NTPLeastSignificant,
76 packet.SR.RTPTimestamp);
197 virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
200 if (parser_->IsRtcp(packet, static_cast<int>(length))) {
202 channel_, packet, static_cast<unsigned int>(length));
205 channel_, packet, static_cas
374 OnSendRtp(const uint8_t* packet, size_t length) argument
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H A Dend_to_end_tests.cc58 virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE {
63 virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
281 virtual Action OnReceiveRtcp(const uint8_t* packet,
283 RTCPUtility::RTCPParserV2 parser(packet, length, true);
286 ssrc |= static_cast<uint32_t>(packet[4]) << 24;
287 ssrc |= static_cast<uint32_t>(packet[5]) << 16;
288 ssrc |= static_cast<uint32_t>(packet[6]) << 8;
289 ssrc |= static_cast<uint32_t>(packet[7]) << 0;
298 << "Timed out while waiting for a receiver RTCP packet to be sent.";
319 virtual Action OnSendRtp(const uint8_t* packet, size_
1125 const RTCPUtility::RTCPPacket& packet = parser.Packet(); local
1129 const RTCPUtility::RTCPPacket& packet = parser.Packet(); local
1185 OnSendRtcp(const uint8_t* packet, size_t length) argument
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H A Drampup_tests.cc106 // Just trigger if there was any rtx padding packet.
116 bool StreamObserver::SendRtp(const uint8_t* packet, size_t length) { argument
119 EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
140 packet,
153 bool StreamObserver::SendRtcp(const uint8_t* packet, size_t length) { argument
265 const uint8_t* packet, size_t length) {
268 EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
279 bool LowRateStreamObserver::SendRtcp(const uint8_t* packet, size_t length) { argument
264 DeliverPacket( const uint8_t* packet, size_t length) argument
H A Dreplay.cc239 test::RtpFileReader::Packet packet; local
240 if (!rtp_reader->NextPacket(&packet))
243 switch (call->Receiver()->DeliverPacket(packet.data, packet.length)) {
249 parser->Parse(packet.data, packet.length, &header);
259 if (last_time_ms != 0 && last_time_ms != packet.time_ms) {
260 SleepMs(packet.time_ms - last_time_ms);
262 last_time_ms = packet.time_ms;
H A Dtransport_adapter.cc20 const void* packet,
25 bool success = transport_->SendRtp(static_cast<const uint8_t*>(packet),
31 const void* packet,
36 bool success = transport_->SendRtcp(static_cast<const uint8_t*>(packet),
19 SendPacket(int , const void* packet, int length) argument
30 SendRTCPPacket(int , const void* packet, int length) argument
H A Dvideo_receive_stream.cc232 bool VideoReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) { argument
234 channel_, packet, static_cast<int>(length)) == 0;
237 bool VideoReceiveStream::DeliverRtp(const uint8_t* packet, size_t length) { argument
239 channel_, packet, static_cast<int>(length), PacketTime()) == 0;
H A Dvideo_send_stream.cc196 // 28 to match packet overhead in ModuleRtpRtcpImpl.
404 bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) { argument
406 channel_, packet, static_cast<int>(length)) == 0;
/external/chromium_org/third_party/webrtc/video_engine/test/auto_test/source/
H A Dvie_autotest_network.cc218 // Create a empty RTP packet.
219 unsigned char packet[3000]; local
220 memset(packet, 0, sizeof(packet));
221 packet[0] = 0x80; // V=2, P=0, X=0, CC=0
222 packet[1] = 0x7C; // M=0, PT = 124 (I420)
224 // Create a empty RTCP app packet.
233 tbChannel.videoChannel, packet, 1500));
238 tbChannel.videoChannel, packet, 1500));
242 tbChannel.videoChannel, packet, 1
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/external/chromium_org/third_party/webrtc/video_engine/test/libvietest/testbed/
H A Dtb_external_transport.cc383 VideoPacket* packet = NULL; local
388 // Take first packet in queue
389 packet = _rtpPackets.front();
391 if (packet)
393 timeToReceive = packet->receiveTime - NowMs();
413 if (packet)
418 ssrc = ((packet->packetBuffer[8]) << 24);
419 ssrc += (packet->packetBuffer[9] << 16);
420 ssrc += (packet->packetBuffer[10] << 8);
421 ssrc += packet
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/external/chromium_org/third_party/webrtc/video_engine/
H A Dvie_receiver.cc202 // Only forward if the incoming packet *and* the channel are both configured
248 // that the first packet is included in the stats).
254 bool ViEReceiver::ReceivePacket(const uint8_t* packet, argument
259 return ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
261 const uint8_t* payload = packet + header.headerLength;
273 bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet, argument
278 if (packet[header.headerLength] == ulpfec_pt)
281 header, packet, packet_length, ulpfec_pt) != 0) {
287 // This is an empty packet and should be silently dropped before trying to
298 LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet
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/external/chromium_org/third_party/webrtc/voice_engine/
H A Dchannel.cc198 "Channel::SendPacket() failed to send RTP packet due to"
206 // Dump the RTP packet to a file (if RTP dump is enabled).
242 "Channel::SendRTCPPacket() failed to send RTCP packet"
250 // Dump the RTCP packet to a file (if RTP dump is enabled).
482 // It is possible that the connection is alive even if no RTP packet has
484 // and a low SID-packet update rate.
523 // packet as discarded.
526 "received packet is discarded since playing is not"
543 // Update the packet delay.
1471 "SetSendCodec() failed to set audio packet siz
1804 ReceivePacket(const uint8_t* packet, int packet_length, const RTPHeader& header, bool in_order) argument
1823 HandleEncapsulation(const uint8_t* packet, int packet_length, const RTPHeader& header) argument
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Completed in 514 milliseconds

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